In Voice-over-IP (VoIP) communications systems, voice signal data is transmitted across a telecommunications network to a receiver as a series of discrete packets. Each packet contains a sample of speech material, each typically comprising one speech “frame,” and the speech material of the transmitted packets is then combined, in sequence, with the other transmitted packets, at the network receiver. (Speech signals are typically divided into a contiguous sequence of “frames,” where each such speech “frame” is a speech segment represents a predetermined time interval, such as, for example, 20 milliseconds.) Thus, the receiver is able to reconstruct the transmitted speech signal for appropriate playback to a listener.
However, since packets cannot be guaranteed to have successfully transited the network, or may not be guaranteed to have done so in an amount of time required for the receiver to reconstruct the speech signal in “real time,” the receiver must somehow have the ability to conceal the effects of packet loss to the user. Such packet loss concealment (PLC) algorithms are well known to those skilled in the art, and typically compensate for “lost” packets by extrapolating from the received speech, in order to generate appropriate “replacement” speech material for the listener. Nearly all conventional PLC algorithms operate based on the assumption that missing speech may be generally well predicted from the immediately preceding speech. This is a typically reasonable assumption because speech tends to be “quasi-stationary” (QS) in nature—that is, the speech signal varies relatively slowly in comparison to the packet size. However, such QS behavior, while usual, does not hold true for all speech packets.
In co-pending U.S. patent application Ser. No. 10/953,907, “Method And Apparatus For Measuring Quality Of Service In Voice-Over-IP Network Applications Based On Speech Characteristics,” filed by M. Lee et. al. on Sep. 29, 2004 and commonly assigned to the assignee of the present invention, a method for measuring the Quality-of-Service (QoS) of a packet-based VoIP communications network is described which is based not only on packet loss rate data (as was conventional), but also on particular characteristics of the speech data itself. In particular, an estimate of what is referred to as “QS failures” (i.e., times when the generally quasi-stationary nature of a speech packet fails) is used therein, in conjunction with a packet loss rate, to calculate the desired QoS measure. In particular, U.S. patent application Ser. No. 10/953,907 introduced the term “voice concealability” to indicate the likelihood that speech will meet the QS assumption of the PLC algorithms, and the term “voice risk” to indicate the likelihood that speech will not meet this QS assumption. Moreover, U.S. patent application Ser. No. 10/953,907 identified certain measures of “voice risk” and “voice concealability” for use in a method for measuring the QoS of a VoIP system. U.S. patent application Ser. No. 10/953,907 is hereby incorporated by reference as if fully set forth herein.
Finally, in order to reduce the required bandwidth, many VoIP systems use a technique known as packet bundling. Packet bundling, familiar to those skilled in the art, occurs when a scheduler or other network element intentionally delays the transmission of some speech frames so that they may be transmitted simultaneously with subsequent speech frames. This advantageously reduces the required bandwidth, since only a single packet header is required to transmit multiple frames, which are thereby transformed into a single packet. However, delaying speech frames for purposes of bundling increases the risk that the receiver will run out of speech material, and therefore will have to run the PLC algorithm, risking quality degradation if the PLC algorithm fails to adequately conceal the lost speech.